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1 // Copyright 2013 The Chromium Authors. All rights reserved. | 1 // Copyright 2013 The Chromium Authors. All rights reserved. |
2 // Use of this source code is governed by a BSD-style license that can be | 2 // Use of this source code is governed by a BSD-style license that can be |
3 // found in the LICENSE file. | 3 // found in the LICENSE file. |
4 | 4 |
5 #include <stddef.h> | 5 #include <stddef.h> |
6 | 6 |
7 #include "base/logging.h" | 7 #include "base/logging.h" |
| 8 #include "base/strings/utf_string_conversions.h" |
| 9 #include "content/renderer/media/mock_media_constraint_factory.h" |
8 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" | 10 #include "content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h" |
| 11 #include "content/renderer/media/webrtc_audio_capturer.h" |
9 #include "content/renderer/media/webrtc_local_audio_source_provider.h" | 12 #include "content/renderer/media/webrtc_local_audio_source_provider.h" |
10 #include "content/renderer/media/webrtc_local_audio_track.h" | 13 #include "content/renderer/media/webrtc_local_audio_track.h" |
11 #include "media/audio/audio_parameters.h" | 14 #include "media/audio/audio_parameters.h" |
12 #include "media/base/audio_bus.h" | 15 #include "media/base/audio_bus.h" |
13 #include "testing/gtest/include/gtest/gtest.h" | 16 #include "testing/gtest/include/gtest/gtest.h" |
14 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" | 17 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" |
15 #include "third_party/WebKit/public/platform/WebString.h" | |
16 #include "third_party/WebKit/public/web/WebHeap.h" | 18 #include "third_party/WebKit/public/web/WebHeap.h" |
17 | 19 |
18 namespace content { | 20 namespace content { |
19 | 21 |
20 class WebRtcLocalAudioSourceProviderTest : public testing::Test { | 22 class WebRtcLocalAudioSourceProviderTest : public testing::Test { |
21 protected: | 23 protected: |
22 void SetUp() override { | 24 void SetUp() override { |
23 source_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | 25 source_params_.Reset(media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
24 media::CHANNEL_LAYOUT_MONO, 48000, 16, 480); | 26 media::CHANNEL_LAYOUT_MONO, 48000, 16, 480); |
25 sink_params_.Reset( | 27 sink_params_.Reset( |
26 media::AudioParameters::AUDIO_PCM_LOW_LATENCY, | 28 media::AudioParameters::AUDIO_PCM_LOW_LATENCY, |
27 media::CHANNEL_LAYOUT_STEREO, 44100, 16, | 29 media::CHANNEL_LAYOUT_STEREO, 44100, 16, |
28 WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize); | 30 WebRtcLocalAudioSourceProvider::kWebAudioRenderBufferSize); |
29 sink_bus_ = media::AudioBus::Create(sink_params_); | 31 sink_bus_ = media::AudioBus::Create(sink_params_); |
| 32 MockMediaConstraintFactory constraint_factory; |
| 33 scoped_refptr<WebRtcAudioCapturer> capturer( |
| 34 WebRtcAudioCapturer::CreateCapturer( |
| 35 -1, StreamDeviceInfo(), |
| 36 constraint_factory.CreateWebMediaConstraints(), NULL, NULL)); |
30 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( | 37 scoped_refptr<WebRtcLocalAudioTrackAdapter> adapter( |
31 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); | 38 WebRtcLocalAudioTrackAdapter::Create(std::string(), NULL)); |
32 scoped_ptr<WebRtcLocalAudioTrack> native_track( | 39 scoped_ptr<WebRtcLocalAudioTrack> native_track( |
33 new WebRtcLocalAudioTrack(adapter.get())); | 40 new WebRtcLocalAudioTrack(adapter.get(), capturer, NULL)); |
34 blink::WebMediaStreamSource audio_source; | 41 blink::WebMediaStreamSource audio_source; |
35 audio_source.initialize(blink::WebString::fromUTF8("dummy_source_id"), | 42 audio_source.initialize(base::UTF8ToUTF16("dummy_source_id"), |
36 blink::WebMediaStreamSource::TypeAudio, | 43 blink::WebMediaStreamSource::TypeAudio, |
37 blink::WebString::fromUTF8("dummy_source_name"), | 44 base::UTF8ToUTF16("dummy_source_name"), |
38 false /* remote */, true /* readonly */); | 45 false /* remote */, true /* readonly */); |
39 blink_track_.initialize(blink::WebString::fromUTF8("audio_track"), | 46 blink_track_.initialize(blink::WebString::fromUTF8("audio_track"), |
40 audio_source); | 47 audio_source); |
41 blink_track_.setExtraData(native_track.release()); | 48 blink_track_.setExtraData(native_track.release()); |
42 source_provider_.reset(new WebRtcLocalAudioSourceProvider(blink_track_)); | 49 source_provider_.reset(new WebRtcLocalAudioSourceProvider(blink_track_)); |
43 source_provider_->SetSinkParamsForTesting(sink_params_); | 50 source_provider_->SetSinkParamsForTesting(sink_params_); |
44 source_provider_->OnSetFormat(source_params_); | 51 source_provider_->OnSetFormat(source_params_); |
45 } | 52 } |
46 | 53 |
47 void TearDown() override { | 54 void TearDown() override { |
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129 // Stop the audio track. | 136 // Stop the audio track. |
130 WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>( | 137 WebRtcLocalAudioTrack* native_track = static_cast<WebRtcLocalAudioTrack*>( |
131 MediaStreamTrack::GetTrack(blink_track_)); | 138 MediaStreamTrack::GetTrack(blink_track_)); |
132 native_track->Stop(); | 139 native_track->Stop(); |
133 | 140 |
134 // Delete the source provider. | 141 // Delete the source provider. |
135 source_provider_.reset(); | 142 source_provider_.reset(); |
136 } | 143 } |
137 | 144 |
138 } // namespace content | 145 } // namespace content |
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